JP3976077B1 - Telephone device - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a low-cost communication apparatus capable of canceling a voice component emitted from a speaker from a transmission signal over a wide frequency band and preventing howling.
SOLUTION: A / D conversion circuits 311 to 31n have elapsed delay times Td1 to Tdn for sound waves in frequency bands G1 to Gn from the timing at which the A / D conversion circuit 301 A / D-converts the audio signal from the microphone M1. Sometimes the audio signal from the microphone M2 is A / D converted, and the output levels of the microphones M1 and M2 are made to coincide with each other in a predetermined frequency band by the attenuation circuits 341 to 34n. Output the signal difference.
[Selection] Figure 1

Description

  The present invention relates to a call device.

  Conventionally, there is a communication device installed indoors with an interphone system or the like, which includes a speaker that outputs sound from a communication device installed in another place, a microphone that inputs sound transmitted to the other communication device, and the like. Yes.

  Since howling occurs when the sound generated from the speaker wraps around the microphone, various measures for preventing howling are taken. For example, a delay circuit that includes a speaker and a pair of microphones, delays the output of a microphone closer to the speaker by a delay time of sound waves corresponding to the difference in distance between the two microphones and the speaker, and both the microphone and the speaker. Adjusting the level corresponding to the difference in distance to match the output level of both microphones with respect to the sound from the speaker, and using both microphone outputs through the delay circuit and the level adjusting amplifier circuit as both inputs There has been proposed a communication device that is provided with a differential amplifier circuit and uses the output of the differential amplifier circuit as a transmission signal.

In this communication device, after picking up the sound from the speakers with both microphones, the delay and the level are adjusted, and the sound components from the speakers input to both microphones are canceled by the differential amplifier circuit. Only the components are removed (cancellation process) to prevent howling. (For example, refer to Patent Documents 1 and 2).
Japanese Patent Laid-Open No. 11-41342 Japanese Patent No. 3226121

  However, the sound emitted from the speaker is transmitted to the microphone and converted into an audio signal, but the transfer function of the sound wave transmitted from the speaker to the microphone has a frequency characteristic, and the phase of the sound collected by the microphone, The amplitude depends on the frequency. Therefore, in the conventional configurations such as Patent Documents 1 and 2, the sound component from the speaker can be canceled near a specific frequency, but the sound component from the speaker cannot be canceled over a wide frequency band. . There is also a demand for cost reduction.

  The present invention has been made in view of the above-mentioned reasons, and its object is to cancel a voice component emitted from a speaker from a transmission signal over a wide frequency band and to prevent a howling from occurring at a low cost. Is to provide.

  According to the first aspect of the present invention, a speaker that outputs transmitted audio information, a first microphone that collects audio and outputs an analog audio signal, and a distance from the speaker that is farther than the first microphone. And a second microphone that collects sound and outputs an analog sound signal, and a signal processing unit that processes and transmits each sound signal output from the first and second microphones. The sound signal transmission time corresponding to the difference in distance between the first and second microphones and the speaker is set as a delay time, and the signal processing unit outputs analog audio signals output from the first and second microphones, respectively. A plurality of either one of the first and second A / D conversion means for converting into a digital signal is provided, and one A / D conversion means corresponds to a predetermined frequency band different from each other, and the other A / D Conversion means Each A / D conversion is performed at a timing that is shifted by a delay time for each predetermined frequency band from the timing at which the A / D conversion is performed, and each output level of the first and second microphones for the sound from the speaker is set to a predetermined frequency band. A level adjusting unit that applies a process for matching each time to the digital signal output from the A / D converting unit, and an arithmetic unit that outputs a difference between the audio signals of the first and second microphones that have passed through the level adjusting unit. It is characterized by.

  According to the present invention, the phase difference and amplitude difference of the outputs of the first and second microphones with respect to the sound from the speaker have frequency dependence, but the sound component emitted by the speaker is canceled for each frequency band from the transmission signal. Thus, howling can be prevented more reliably. Further, even when A / D conversion is performed at low speed, the delay time between the audio signals of the first and second microphones with respect to the sound emitted from the speaker can be accurately compensated, and the A / D conversion means can be operated at high speed. Since what can be processed is not necessary, the cost can be reduced. That is, it is possible to provide a low-cost communication device that can cancel the sound component emitted from the speaker from the transmitted signal over a wide frequency band and prevent the occurrence of howling.

  According to a second aspect of the present invention, in the first aspect, the signal processing unit includes a first filter unit that passes a predetermined frequency band for each output of the one A / D conversion unit, and the other A / D conversion. And second filter means for separating and passing the output of the means for each predetermined frequency band, and the computing means passes through the first and second filter means and the level adjusting means. A difference between the audio signals of the microphones for each predetermined frequency band is added.

  According to the present invention, the phase difference and amplitude difference of the outputs of the first and second microphones with respect to the sound from the speaker have frequency dependence, but the sound component emitted by the speaker is canceled for each frequency band from the transmission signal. Thus, howling can be prevented more reliably.

  According to a third aspect of the present invention, in the first or second aspect, the signal processing unit includes one first A / D conversion unit, a plurality of the second A / D conversion units, and a plurality of the second A / D conversion units. Each of the second A / D conversion means corresponds to each predetermined frequency band, and each of the second A / D conversion means has a timing after a delay time for each predetermined frequency band from the timing when the first A / D conversion means performs A / D conversion. A first filter means that performs A / D conversion and further separates and passes the output of the first A / D conversion means for each predetermined frequency band, and predetermined for each output of the second A / D conversion means And a second filter means for passing the frequency band, wherein the computing means has a predetermined frequency of the audio signals of the first and second microphones that have passed through the first and second filter means and the level adjusting means. The difference between the bands is added.

  According to the present invention, the phase difference and amplitude difference of the outputs of the first and second microphones with respect to the sound from the speaker have frequency dependence, but the sound component emitted by the speaker is canceled for each frequency band from the transmission signal. Thus, howling can be prevented more reliably.

  According to a fourth aspect of the present invention, in the first or second aspect, the signal processing unit includes a plurality of the first A / D conversion units, a single second A / D conversion unit, and a plurality of the first A / D conversion units. Each of the first A / D conversion means corresponds to a predetermined frequency band, and each of the first A / D conversion means has a timing before a delay time for each predetermined frequency band from a timing at which the second A / D conversion means performs A / D conversion. A first filter means for performing A / D conversion and passing a predetermined frequency band for each output of the first A / D conversion means; and an output of the second A / D conversion means for each predetermined frequency band And a second filter means that passes the first and second filter means and the level adjusting means, and the arithmetic means has a predetermined frequency of the audio signals of the first and second microphones that have passed through the first and second filter means. The difference between the bands is added.

  According to the present invention, the phase difference and amplitude difference of the outputs of the first and second microphones with respect to the sound from the speaker have frequency dependence, but the sound component emitted by the speaker is canceled for each frequency band from the transmission signal. Thus, howling can be prevented more reliably.

  As described above, according to the present invention, there is an effect that it is possible to provide a low-cost communication device that can cancel the sound component emitted from the speaker from the transmission signal over a wide frequency band and prevent the occurrence of howling. is there.

  Hereinafter, embodiments of the present invention will be described with reference to the drawings.

(Embodiment 1)
The communication device A of the present embodiment is shown in FIGS. 2 to 4, and a housing A1 is configured by a body A10 having an opening formed on the rear surface and a cover A11 covering the opening of the body A10. A speaker SP, a microphone board MB1, a call switch SW1, and a voice processing unit 10 are provided.

  As shown in FIG. 4, the audio processing unit 10 includes an IC including a communication unit 10a, echo cancellation units 10b and 10c, an amplification unit 10d, and a signal processing unit 10e, and is disposed in the housing A1. A voice signal transmitted from the communication device A installed in another room or the like via the information line Ls is received by the communication unit 10a, amplified by the amplification unit 10d via the echo cancellation unit 10b, and then the speaker. Output from SP. Further, by operating the call switch SW1, a call can be made and each audio signal input from the microphone M1 (first microphone) and the microphone M2 (second microphone) on the microphone board MB1 is received by the signal processing unit 10e. After being subjected to signal processing to be described later, the signal passes through the echo cancel unit 10c, and is transmitted from the communication unit 10a to the communication device A installed in another room or the like via the information line Ls. That is, it functions as an intercom that allows two-way calls between rooms. Note that the power of the communication device A may be supplied from an outlet provided in the vicinity of the installation location or may be supplied via the information line Ls.

  As shown in FIG. 2, the speaker SP is a cylindrical yoke 20 formed of an iron-based material having a thickness of about 0.8 mm such as cold rolled steel plate (SPCC, SPCEN), electromagnetic soft iron (SUY), etc., and having one end opened. The circular support body 21 is extended from the opening end of the yoke 20 toward the outside.

  A cylindrical permanent magnet 22 (for example, residual magnetic flux density of 1.39 T to 1.43 T) formed of neodymium is disposed in the cylinder of the yoke 20, and the outer peripheral edge of the dome-shaped diaphragm 23 is a support. 21 is fixed to the edge surface.

  The diaphragm 23 is formed of a thermoplastic plastic (for example, a thickness of 12 μm to 50 μm) such as PET (PolyEthyleneTerephthalate) or PEI (Polyetherimide). A cylindrical bobbin 24 is fixed to the rear surface of the diaphragm 23, and is formed by winding a polyurethane copper wire (for example, φ0.05 mm) around a paper tube of kraft paper at the rear end of the bobbin 24. A voice coil 25 is provided. The bobbin 24 and the voice coil 25 are provided so that the voice coil 25 is positioned at the opening end of the yoke 20, and freely move in the front-rear direction in the vicinity of the opening end of the yoke 20.

  When an audio signal is input to the polyurethane copper wire of the voice coil 25, an electromagnetic force is generated in the voice coil 25 due to the current of the audio signal and the magnetic field of the permanent magnet 22, so that the bobbin 24 moves back and forth with the diaphragm 23. Visible in the direction. At this time, a sound corresponding to the audio signal is emitted from the diaphragm 23. That is, an electrodynamic speaker SP is configured.

  And the rib 11 is formed in the front inner side of housing A1 which the diaphragm 23 of speaker SP opposes, and the end surface of the convex part 21a protruded to the front side from the outer peripheral end part of the circular support body 21 of speaker SP. Comes into contact with the rib 11, and the speaker SP is fixed in a state where the diaphragm 23 faces the front surface of the housing A1 from the inside.

  When the speaker SP is fixed in the housing A1, the front air chamber Bf which is a space surrounded by the front inner side of the housing A1 and the front surface side (the diaphragm 23 side) of the speaker SP, the rear inner side and the inner side surface of the housing A1. And a rear air chamber Br which is a space surrounded by the back surface side (yoke 20 side) of the speaker SP. The front air chamber Bf communicates with the outside through a plurality of sound holes 12 provided on the front surface of the housing A1. The rear air chamber Br becomes a space that is insulated (not communicated) with the front air chamber Bf by closely contacting the end portion of the support 21 of the speaker SP and the rib 11 on the inner surface of the housing A1, and further covers the cover A11. Is in close contact with the rear opening of the body A10 to form a sealed space that is insulated from the outside.

  Next, as shown in FIG. 5, the microphone substrate MB1 has a pair of microphone bare chips BC1 and ICKa1 and a pair of microphone bare chips BC2 and ICKa2 mounted on one surface 2a of the module substrate 2, respectively, and bare chips BC1 and ICKa1. After connecting the wiring patterns (not shown) on the module substrate 2 and between the bare chips BC2, ICKa2 and the wiring patterns (not shown) on the module substrate 2 with wires W (wire bonding), A shield case SC1 is mounted so as to cover the pair of bare chips BC1 and ICKa1, and a shield case SC2 is mounted so as to cover the pair of bare chips BC2 and ICKa2, so that the microphone configured by the bare chips BC1, ICKa1, and shield case SC1 M1, bare chip BC2, I Ka2, and a composed microphone M2 with a shield case SC2.

  As shown in FIG. 6, in the bare chip BC (bare chip BC1 or BC2), an Si thin film 1d is formed on one surface side of the silicon substrate 1b so as to close the hole 1c formed in the silicon substrate 1b. An electrode 1f is formed between them via an air gap 1e, and a pad 1g for outputting an audio signal is further provided to constitute a capacitor type silicon microphone. Then, an external acoustic signal vibrates the Si thin film 1d, whereby the capacitance between the Si thin film 1d and the electrode 1f changes to change the amount of charge, and the pad 1g changes with this change in the amount of charge. , 1g, a current corresponding to the acoustic signal flows. In this bare chip BC, the silicon substrate 1b is die-bonded on the module substrate 2. In particular, the Si thin film 1d of the bare chip BC2 faces the sound hole F2 formed in the module substrate 2.

  The microphone M1 uses the bottom surface side of the shield case SC1 with the sound hole F1 as a sound collecting surface, and the microphone M2 uses the mounting surface side with respect to the module substrate 2 with the sound hole F2 as a sound collecting surface. The module substrate 2 has sound collecting surfaces in both directions opposite to each other. Since the microphone substrate MB1 configured in this manner has both the microphones M1 and M2 mounted on the one surface 2a of the module substrate 2, the thickness of the microphone substrate MB1 can be reduced.

  FIG. 7A is a plan view of the microphone board MB1 as viewed from the one surface 2a side of the module board 2. The module board 2 has a rectangular part 2f in which the microphone M1 is arranged and a rectangular part 2g in which the microphone M2 is arranged. And a connecting portion 2h that connects the rectangular portions 2f and 2g, and the rectangular portion 2g is formed larger than the rectangular portion 2f. A negative power supply pad P1, a positive power supply pad P2, an output 1 pad P3, and an output 2 pad P4 are provided along the edge of the rectangular portion 2g.

  As shown in FIG. 7B, the negative power supply pad P1 is connected to the negative side of the power supply voltage supplied from the outside, and the positive power supply pad P2 is connected to the positive side of the power supply voltage. Power is supplied to the microphones M1 and M2 through the pattern. Further, an audio signal collected by the microphone M1 is output from the output 1 pad P3 via a wiring pattern on the module substrate 2, and an audio signal collected by the microphone M2 is output from the output 2 pad P4. It is output via the upper wiring pattern. The ground of the audio signal output from the output pads P3 and P4 is shared by the negative power supply pad P1.

  Thus, the power of the microphones M1 and M2 is supplied from the common negative power supply pad P1 and the positive power supply pad P2, and the ground of each output of the microphones M1 and M2 is shared by the negative power supply pad P1, so that the number of pads The configuration can be simplified.

  Next, the operation of the microphone substrate MB1 will be described.

  First, each current flowing from the bare chips BC1 and BC2 according to the collected acoustic signal is impedance-converted by ICKa1 and Ka2 and converted into a voltage signal, and output from the output 1 pad P3 and the output 2 pad P4 as audio signals, respectively. Is done.

  The ICKa (ICKa1 or Ka2) has the circuit configuration of FIG. 8, and is a constant IC composed of a chip IC that converts the power supply voltage + V (for example, 5V) supplied from the power supply pads P1 and P2 into a constant voltage Vr (for example, 12V). A voltage circuit Kb is provided, a constant voltage Vr is applied to the series circuit of the resistor R11 and the bare chip BC, and a connection midpoint between the resistor R11 and the bare chip BC is connected to the junction type J-FET element S11 via the capacitor C11. Connected to the gate terminal. The drain terminal of the J-FET element S11 is connected to the operating power supply + V, and the source terminal is connected to the negative side of the power supply voltage via the resistor R12. Here, the J-FET element S11 is for electrical impedance conversion, and the voltage at the source terminal of the J-FET element S11 is output as an audio signal. Note that the ICKa impedance conversion circuit is not limited to the above-described configuration, and may be, for example, a circuit having a function of a source follower circuit using an operational amplifier, or an audio signal amplification circuit in the ICKa if necessary. May be provided.

  The microphone board MB1 can efficiently configure the signal lines and the power supply lines by performing signal transmission and power supply via the wiring pattern on the module board 2 as described above, and can be attached to the outer surface of the housing A1. Become. In this embodiment, one surface 2a of the module substrate 2 is arranged along the outer front surface of the housing A1, and the microphone M1 is inserted through the opening 13 on the front surface of the housing A1 so that the sound collection surface faces the front air chamber Bf, and the shield The sound hole F1 of the microphone M1 formed in the bottom surface of the case SC1 faces the diaphragm 23 of the speaker SP, and the sound emitted from the speaker SP can be reliably collected through the sound hole F1. The microphone M2 is fitted into a recess 14 provided in the front surface of the housing A1, and the sound hole F2 of the microphone M2 formed in the module substrate 2 is located outside (frontward) the housing A1 in the output direction of the speaker SP. Therefore, it is possible to reliably collect the sound transmitted from the speaker located in front of the communication device A and transmitted through the sound hole F2. If the distances from the center of the speaker SP to the centers of the microphones M1 and M2 are X1 and X2, respectively, X1 <X2.

  Further, the rear air chamber Br facing the back surface of the speaker SP is sealed in the housing A1, so that sound radiated from the back surface of the speaker SP is difficult to leak from the rear air chamber Br, and the speaker SP and the microphone M2 are not connected. Acoustic coupling is reduced. Furthermore, the sound radiated from the back surface of the speaker SP (the back surface of the diaphragm 23) has a phase reversed from that of the sound radiated from the surface of the speaker SP (the surface of the diaphragm 23). When the generated sound circulates forward, the sound radiated from the surface of the speaker SP cancels each other, the radiated sound pressure of the speaker SP decreases, and the speaker in front cannot hear the sound emitted by the speaker SP. However, since the sound radiated from the back surface of the speaker SP is difficult to leak to the outside of the housing A1 as described above, a decrease in the radiated sound pressure of the speaker SP due to the wraparound is prevented.

  Further, since the concave portion 14 in which the microphone M2 is accommodated is a separated space that does not communicate with the rear air chamber Br, the microphone M2 is more difficult to collect the sound emitted by the speaker SP, and the speaker SP and the microphone M2 are separated. The acoustic coupling is further reduced. That is, with the above configuration, the sound emitted from the speaker SP and the sound emitted from the speaker are separated and collected by the microphones M1 and M2.

  Further, when the microphone substrate MB1 is disposed in the housing A1, it is difficult to maintain the spatial insulation between the front air chamber Bf and the rear air chamber Br, but the microphone substrate MB1 is disposed in the housing as in the present embodiment. By attaching to the outer surface of A1, the spatial insulation between the front air chamber Bf and the rear air chamber Br can be maintained.

  And in this embodiment, in order to prevent the howling which generate | occur | produces when the microphones M1 and M2 pick up the audio | voice output of the speaker SP, it has the following structures.

  First, as shown in FIG. 1, the signal processing unit 10 e housed in the audio processing unit 10 includes one A / D conversion circuit 301, a plurality of A / D conversion circuits 311 to 31 n, and a bandpass filter 321. To 32n, bandpass filters 331 to 33n, attenuation circuits 341 to 34n, an arithmetic circuit 35a, and a timing control unit 36. Each operation of the A / D conversion circuits 301 and 311 to 31n is controlled by the timing control unit 36, and the operation of each circuit will be described below. 9 to 12 show signal waveforms in each part of the signal processing unit 10e when the microphones M1 and M2 collect the sound emitted from the speaker SP.

  First, the microphone M1 has a sound collection surface mounted toward the speaker SP, and the speaker SP is more effective than the sound (transmitted sound) emitted by the speaker H (see FIG. 1) located in front of the call device A. The sound that is emitted is collected with high sensitivity. On the other hand, the microphone M2 has a sound collection surface arranged forward, and collects sound emitted by the speaker H located in front of the communication device A with higher sensitivity than the sound emitted by the speaker SP.

  That is, the sound signal Y11 output from the microphone M1 has high sensitivity and large amplitude with respect to the sound emitted from the speaker SP, but has low sensitivity and small amplitude with respect to the sound emitted from the speaker H. The sound signal Y21 output from the microphone M2 has high sensitivity and large amplitude for the sound emitted by the speaker H, but has low sensitivity and small amplitude for the sound emitted from the speaker SP.

  Furthermore, at the time of transmission, both the sound emitted by the speaker H and the sound emitted by the speaker SP are collected by the microphones M1 and M2, but the distance X1 from the center of the speaker SP to the center of each of the microphones M1 and M2 , X2 satisfy X1 <X2, so that a phase difference occurs between the sound signal Y11 of the microphone M1 and the sound signal Y21 of the microphone M2 with respect to the sound from the speaker SP, and both the microphones M1, M2 and the speaker SP. The sound signal Y21 of the microphone M2 is compared with the sound signal Y11 of the microphone M1 by the sound wave delay time [Td = (X2-X1) / Vs] (Vs is the speed of sound) corresponding to the difference (X2-X1) in distance from the sound wave. Is out of phase. The delay time Td is constant with respect to the frequency without depending on the frequency of the sound emitted from the speaker SP under ideal conditions (for example, a point sound source, a sealed housing structure, and no variation in CR of the circuit configuration). Although there is actually no ideal condition, it depends on the frequency of the sound emitted by the speaker SP.

  Similarly, the amplitudes of the audio signals Y11 and Y21 with respect to the sound from the speaker SP are constant with respect to the frequency under the ideal conditions without depending on the frequency of the sound emitted from the speaker SP. Since it is not an ideal condition, it depends on the frequency of the sound emitted by the speaker SP.

  On the other hand, since the distances between the microphones M1 and M2 and the speaker H can be considered to be equal, the voice signals Y11 and Y21 of the microphones M1 and M2 have substantially the same phase with respect to the voice emitted by the speaker H. .

  Thus, in the present embodiment, the audio signal from the microphones M1 and M2 is divided for each frequency band, and the phase difference and amplitude difference generated between the audio signal Y11 of the microphone M1 and the audio signal Y21 of the microphone M2 are determined. It is optimally adjusted for each frequency band. 9A to 9H show the operation when the frequency band division number n = 2.

  First, the A / D conversion circuit 301 converts the analog audio signal Y11 of the microphone M1 into a digital signal, and the A / D conversion circuits 311 to 31n convert the analog audio signal Y21 of the microphone M2 into a digital signal. The A / D conversion operation of the A / D conversion circuit 301 is performed in synchronization with the rising edge of the control signal Si11 output from the timing control unit 36, and the A / D conversion operations of the A / D conversion circuits 311 to 31n are the timing. It is performed in synchronization with the rise of the control signals Si21 to Si2n output from the control unit 36, and the frequency of the control signals Si11 and Si21 to Si2n in this embodiment is set to 8 to 16 KHz, and each A / D conversion circuit The conversion cycle T1 is 62.5 to 125 μsec (FIGS. 9C to 9E).

  Further, when the frequency band of the sound emitted from the speaker SP is divided into n parts G1 to Gn, the phase of the sound signal Y21 of the microphone M2 is the sound signal of the microphone M1 in the lowest frequency band G1 with respect to the sound emitted from the speaker SP. Delayed by a delay time Td11 compared to Y11, and in the next lower frequency band G2, the phase of the audio signal Y21 of the microphone M2 is delayed by a delay time Td12 compared to the audio signal Y11 of the microphone M1,..., Nth lowest frequency In the band Gn, the phase of the audio signal Y21 of the microphone M2 is delayed by a delay time Td1n compared to the audio signal Y11 of the microphone M1. Note that Td11> Td12>...> Td1n.

  Therefore, the control signal Si21 is generated later than the control signal Si11 by the delay time Td11, the control signal Si22 is generated later than the control signal Si11 by the delay time Td12,..., And the control signal Si2n is generated in the control signal Si11. On the other hand, the operation of the timing control unit 36 is set in advance so as to be delayed by the delay time Td1n (see FIGS. 9A and 9B). That is, the A / D conversion circuit 311 performs A / D conversion on the audio signal from the microphone M2 at a timing after the delay time Td11 from the timing at which the A / D conversion circuit 301 performs A / D conversion on the audio signal from the microphone M1. The A / D conversion circuit 312 performs A / D conversion on the audio signal from the microphone M2 at a timing after the delay time Td12 from the timing at which the A / D conversion circuit 301 performs A / D conversion on the audio signal from the microphone M1. The A / D conversion circuit 31n receives the audio signal from the microphone M2 at the timing after the delay time Td1n from the timing at which the A / D conversion circuit 301 converts the audio signal from the microphone M1 into the A / D conversion circuit. By performing D conversion, the phases of the audio signal of the microphone M1 and the audio signal of the microphone M2 are converted into frequency bands G1 to Gn. It is made to coincide to.

  Further, the A / D conversion circuit 301 and the A / D conversion circuits 311 to 31n perform A / D conversion by the ΣΔ method, and change the frequency transfer characteristic of only noise, which is a feature of the ΣΔ method, to reduce the noise component. Noise shaping distributed in a higher frequency region than the signal component enables A / D conversion with a high S / N ratio, and further, most of the processing is performed digitally, which facilitates IC integration. Note that the resolution of A / D conversion is any one of 16 bits, 14 bits, and 12 bits.

  The A / D conversion circuit 301 outputs an audio signal Y12 composed of digital values a11, b11, c11, d11, e11,... Obtained by A / D converting the audio signal output from the microphone M1. (Refer FIG.9 (f)). The A / D conversion circuit 311 outputs an audio signal Y221 composed of digital values a21, b21, c21, d21, e21,... Obtained by A / D converting the audio signal output from the microphone M2. (See FIG. 9 (h)). From the A / D conversion circuit 312, the sound is composed of digital values a22, b22, c22, d22, e22,... A / D converted from the sound signal output from the microphone M2. The signal Y222 is output (see FIG. 9 (g)),..., From the A / D conversion circuit 31n, digital values a2n, b2n, c2n, d2n, A / D converted audio signals output from the microphone M2. An audio signal Y22n composed of e2n,... is output, and the audio signal of the microphone M2 is an n-system sound that is phase-adjusted corresponding to each of the frequency bands G1 to Gn. The signal Y221~Y22n.

  In this way, the A / D conversion is performed at the timing at which the phases of the sound signal of the microphone M1 and the sound signal of the microphone M2 coincide with each other for each of the frequency bands G1 to Gn. Even when A / D conversion is performed at 16 KHz), the delay times Td1 to Tdn between the sound signals of the microphones M1 and M2 with respect to the sound emitted from the speaker SP are accurately compensated, and a high-speed A / D conversion function (for example, 1 MHz) The cost can be reduced. In the following processing, the digital values of the audio signals of the microphones M1 and M2 are stored in the memory and digitally processed. For the sake of explanation, the waveforms in FIGS. 10 to 12 are shown as analog waveforms. .

  Next, in the audio signal Y12 output from the A / D conversion circuit 301, the component of the frequency band G1 is separated by the band pass filter 321, and the component of the frequency band G2 is separated by the band pass filter 322,. The components of the frequency band Gn are separated by the band pass filter 32n and distributed to n systems. The audio signal Y 221 output from the A / D conversion circuit 311 passes the component of the frequency band G 1 by the band pass filter 331, and the audio signal Y 222 output from the A / D conversion circuit 312 is the band pass filter 332. The component of the frequency band G2 passes through the sound signal Y22n output from the A / D conversion circuit 31n, and the component of the frequency band Gn passes through the bandpass filter 33n. The audio signals Y131 and Y231 in the frequency band G1 output from the bandpass filters 321 and 331 have the same phase by operating the A / D conversion circuit 301 and the A / D conversion circuit 311 with the delay time Td11 shifted from each other. The audio signals Y132 and Y232 in the frequency band G2 output from the bandpass filters 322 and 332 operate the A / D conversion circuit 301 and the A / D conversion circuit 312 with the delay time Td12 shifted from each other. ,..., The audio signals Y13n and Y23n in the frequency band Gn output from the bandpass filters 32n and 33n are delayed from each other by the A / D conversion circuit 301 and the A / D conversion circuit 31n. The same phase is obtained by operating by shifting the time Td1n (see FIG. 10).

  Next, the attenuation circuit 341 generates an audio signal Y141 by attenuating the audio signal Y131 of the microphone M1 that has passed through the bandpass filter 321, and the attenuation circuit 342 outputs the audio signal Y132 of the microphone M1 that has passed through the bandpass filter 322. Is attenuated to generate the audio signal Y142, and the attenuation circuit 34n attenuates the audio signal Y13n of the microphone M1 that has passed through the bandpass filter 32n to generate the audio signal Y14n (see FIG. 11). For each frequency band G1 to Gn, the level adjustment corresponding to the difference in distance between the microphones M1, M2 and the speaker SP (X2-X1) and the sensitivity difference between the microphones M1, M2 is performed, and the sound emitted from the speaker SP is adjusted. Thus, the output levels of the microphones M1 and M2 in the frequency bands G1 to Gn are matched.

  Next, the arithmetic circuit 35a subtracts the sound signal Y141 of the microphone M1 from the sound signal Y231 of the microphone M2, thereby generating a sound signal in which the sound component from the speaker SP is canceled in the frequency band G1, and the microphone M2 By subtracting the audio signal Y142 of the microphone M1 from the audio signal Y232, an audio signal in which the audio component from the speaker SP is canceled in the frequency band G2 is generated,..., ... from the audio signal Y23n of the microphone M2 By subtracting the audio signal Y14n, an audio signal in which the audio component from the speaker SP is canceled in the frequency band Gn is generated (see FIG. 12A), and the subtraction results in the frequency bands G1 to Gn are further obtained. All are added, and the sound component from the speaker SP is in each frequency band. Audio signal Ya that is canceled for each 1~Gn is generated (see FIG. 12 (b)). In this embodiment, the arithmetic circuit 35a subtracts the sound signal of the microphone M1 from the sound signal of the microphone M2. However, the sound signal Ya is generated by subtracting the sound signal of the microphone M2 from the sound signal of the microphone M1. May be.

  On the other hand, for the sound uttered by the speaker H in front of the microphones M1 and M2, the amplitude of the sound signal Y21 of the microphone M2 having the sound collection surface arranged toward the speaker H is such that the sound collection surface faces the speaker SP. It becomes larger than the amplitude of the audio signal Y11 of the arranged microphone M1. Further, since the signal from the microphone M1 is attenuated by the attenuation circuits 341 to 34n, the speech component from the speaker H included in the speech signals Y231 to 23n is the speech component from the speaker H included in the speech signals Y141 to 14n. Even bigger. That is, the amplitude difference between the speech component from the speaker H included in the speech signals Y141 to 14n and the speech component from the speaker H included in the speech signals Y231 to Y23n becomes large, and the subtracting process is performed by the arithmetic circuit 35a. Even if applied, the signal corresponding to the voice uttered by the speaker H remains in the voice signal Ya in a state where the amplitude is maintained sufficiently.

  In the audio signal Ya output from the signal processing unit 10e as described above, the audio component from the speaker SP is reduced for each of the frequency bands G1 to Gn, while the speaker H in front of the communication device A is directed to the microphones M1 and M2. In the audio signal Ya, the relative difference between the audio component from the speaker H to be kept and the audio component from the speaker SP to be reduced becomes large over a wide frequency band. The effect of preventing howling that occurs when the microphones M1 and M2 pick up the sound output of the speaker SP is improved.

  Next, a process for reducing the sound from the speaker picked up by the microphone by dividing the sound signal of the microphone for each frequency band as in this embodiment will be described using mathematical expressions.

  First, when the microphone M1 in which the sound collection surface is arranged toward the speaker SP and the microphone M2 in which the sound collection surface is arranged toward the speaker H respectively collect the sounds from the speaker SP, the microphones M1 and M2 Assuming that the phase difference and amplitude difference generated between the audio signals do not depend on the frequency, Q1 (ω) is the sound collection characteristic of the microphone M1, and Q2 ′ (ω) is the sound collection characteristic of the microphone M2.

It becomes. Here, α, α ₁ , and β are constants, each first term of [Equation 1] and [Equation 2] is a signal from the speaker SP (speaker sound), and each second term is a speaker H It is a signal by voice.

  Then, without considering the frequency dependence of the phase difference and the amplitude difference, a canceling process for reducing the sound from the speaker SP by subtracting Q2 ′ (ω) from the value obtained by multiplying the attenuation factor a by Q1 (ω). If done, the signal Ycan 'after cancellation is

Thus, the signal from the speaker SP is canceled and only the signal from the speaker H remains.

  However, in reality, the phase difference and amplitude difference generated between the audio signals of the microphones M1 and M2 depend on the frequency, and the sound collection characteristic Q2 (ω) of the microphone M2 in consideration of this frequency dependency is

It becomes.

  Then, in consideration of the frequency dependence of the phase difference and the amplitude difference, canceling processing for reducing the sound from the speaker SP by subtracting Q2 (ω) from the value obtained by multiplying the attenuation factor a by Q1 (ω) is performed. The canceled signal Ycan is

It becomes. Here, the first term of [Equation 5] is a residual signal of the sound from the speaker SP, and θ (ω) in the first term is the sound when the sound from the speaker SP is collected by the microphones M1 and M2. This is the phase difference (microphone M2−microphone M1) and is hereinafter referred to as the phase difference θ (ω) of the speaker sound. The second term is a voice signal from the speaker H.

Next, {α ₁ (ω) exp (−jθ (ω)) − α ₁ } in the first term of [Equation 5], which is a residual signal of speech from the speaker SP, is expanded for each frequency band {α Focusing on ₁ (ω) exp (−jθ _k (ω)) − α _k }, the problem is how much the amplitude of the signal from the speaker SP is attenuated from α _1. If α ₁ (ω) and α _k divided by α are a (ω) and a _k , respectively, [Equation 6] is obtained.

Here, a (ω) is the amplitude ratio (microphone M2 / microphone M1) of each audio signal obtained by collecting the sound from the speaker SP with the microphones M1 and M2, and is hereinafter referred to as the amplitude ratio a (ω) of the speaker sound. Called. Then, | a (ω) exp {−j (θ _k (ω) −θ)} − a _k | in [Equation 6] is _represented by a phasor vector as a vector Ve in FIG.

  Next, the actually measured speaker sound amplitude ratio a (ω) and the speaker sound phase difference θ (ω) are shown in FIGS. Both the amplitude ratio a (ω) of the speaker sound and the phase difference θ (ω) of the speaker sound tend to increase as the frequency increases. Therefore, the amplitude ratio a (ω) of the speaker sound changes linearly from −20 dB (0.1) to −16 dB (0.16) at a frequency of 400 Hz to 4000 Hz, and the phase difference θ (ω) of the speaker sound is changed. Assuming that the frequency changes linearly from −5 ° to 25 ° at a frequency of 400 Hz to 4000 Hz, approximate expressions of [Equation 7] and [Equation 8] are derived.

  The following [Equation 9] is a mathematical formula for deriving the cancellation amount Zm of the speaker sound.

Using the formula obtained by substituting the above [Equation 7] and [Equation 8] into this [Equation 9], the amount of cancellation of the speaker sound is optimal at one frequency of 1 KHz or 3 KHz without performing frequency division of this embodiment. The theoretical values Yt1 and Yt2 of the cancellation amount of the speaker sound when adjusted to be as shown in FIG. 15A are shown in FIG. 15A, and the actually measured values Ys1 and Ys2 of the cancellation amount of the speaker sound in this case are shown in FIG. ) And is represented by the characteristic that the cancellation amount peaks at a frequency of 1 KHz or 3 KHz, and the frequency range in which the speaker sound canceling effect is sufficiently obtained in each characteristic is relatively centered on the frequency of 1 KHz or 3 KHz. Limited to a narrow range. Note that the amount of cancellation of speaker sound is the amount of reduction in speaker sound that has sneak into the microphone M2 for the purpose of collecting the sound from the speaker H.

  On the other hand, the frequency division of the present embodiment is performed using the formula obtained by substituting the above [Equation 7] and [Equation 8] into the above [Equation 9], and the cancellation amount of the speaker sound is optimized at a frequency of 1 KHz, which is 2 KHz or less. The theoretical value Yt11 of the cancellation amount of the speaker sound when the frequency band G1 and the cancellation amount of the speaker sound are divided into two of the frequency band G2 of 2 KHz or more optimized at the frequency of 3 KHz is shown in FIG. The measured value Ys11 of the amount of cancellation of the speaker sound when the signal processing unit 10e of the embodiment is adjusted to be optimal in the frequency band G1 of 2 KHz or less and the frequency band G2 of 2 KHz or more is shown in FIG. As shown in (b), the cancellation amount peaks at both frequencies of 1 KHz and 3 KHz. It can be seen that there is Nseru effect.

  Note that the theoretical value Yt11 of the speaker sound cancellation amount shown in FIG. 16A and the actual measurement value Ys11 of the speaker sound cancellation amount shown in FIG. The frequency characteristics are improved as theoretically, but the amount of cancellation is different. This is because the actual measurement environment is (1) the S / N ratio of the microphones M1 and M2 actually used is about 60 dB, and (2) the sound of the sound that wraps around the speaker SP in a place where the background noise is about 40 dBA. The reason is considered to be that the pressure is about 80 dB, (3) there are various noises on the actual machine, and (4) there is a resolution limit due to the quantization bit at the time of A / D conversion.

  The audio signal Ya output from the signal processing unit 10e is output to the echo cancellation unit 10c, and the echo cancellation units 10b and 10c (see FIG. 4) perform further processing to prevent further howling.

  First, the echo canceling unit 10c captures the output of the echo canceling unit 10b as a reference signal, and performs an operation on the output of the signal processing unit 10e, thereby further processing the audio signal that has circulated from the speaker SP to the microphones M1 and M2. Cancel. On the other hand, the echo canceling unit 10b also captures the output of the echo canceling unit 10c as a reference signal and performs an operation on the output of the communication unit 10a, thereby wrapping the audio signal from the speaker to the microphone on the other party side Cancel.

  Specifically, the echo cancellation units 10b and 10c are configured by a speaker SP-microphone M1, M2-signal processing unit 10e-echo cancellation unit 10c-communication unit 10a-echo cancellation unit 10b-amplification unit 10d-speaker SP. By adjusting the amount of loss in a variable loss means (not shown) provided in the loop circuit, the loop gain is set to 1 or less to prevent howling. Here, it is assumed that the smaller signal level of the transmission signal and the reception signal is not important, and the transmission loss of the variable loss circuit inserted in the transmission line having the smaller signal level is increased.

  Further, the number of microphones M1 and M2 is not limited to one each, and a plurality of microphones may be provided as the microphones M1 and M2.

  In the present embodiment, one A / D conversion circuit is provided on the microphone M1 side, a plurality of A / D conversion circuits are provided on the microphone M2, and the A / D conversion circuit on the microphone M1 side performs A / D conversion. The A / D conversion circuit on the microphone M2 side performs A / D conversion at the timing after the delay times Td1 to Tdn for the frequency bands G1 to Gn from the timing of performing the audio signal, so that the audio signal of the microphone M1 and the microphone M2 Each phase of the audio signal is matched for each of the frequency bands G1 to Gn. However, contrary to this configuration, a plurality of A / D conversion circuits are provided on the microphone M1 side, one A / D conversion circuit is provided on the microphone M2, and the A / D conversion circuit on the microphone M2 side is an A / D conversion circuit. The A / D conversion circuit on the microphone M1 side performs A / D conversion at the timing before the delay times Td1 to Tdn for each frequency band G1 to Gn from the timing of performing D conversion, and the above filter processing is performed on these digital signals. Similar effects can be obtained by performing arithmetic processing.

(Embodiment 2)
In the signal processing unit 10e of the first embodiment, band division filters 321 to 32n and 331 to 33n are provided in the previous stage of the arithmetic circuit 35a to perform frequency division. However, in this embodiment, the filter unit is not provided and the embodiment is not provided. A signal processing unit 10e that can obtain substantially the same effect as the frequency division of 1 will be described. In addition, the same code | symbol is attached | subjected to the structure similar to Embodiment 1, and description is abbreviate | omitted.

  First, the signal processing unit 10e of the present embodiment includes an arithmetic circuit 35b as shown in FIG. 17, and the arithmetic circuit 35b receives the audio signal Y12 from the A / D conversion circuit 301 through the attenuation circuits 341 to 34n. The signals Y141 to 14n are input, and the audio signals Y221 to 22n from the A / D conversion circuits 311 to 31n are directly input. That is, by setting the A / D conversion timings of the A / D conversion circuits 301 and 311 to 31n for the frequency bands G1 to Gn in the same manner as in the first embodiment, the microphone M1, The M2 audio signals have the same phase for each of the frequency bands G1 to Gn, and the attenuation circuits 341 to 34n make the audio signals of the microphones M1 and M2 have the same amplitude for each of the frequency bands G1 to Gn. The signals are input to the arithmetic circuit 35b without passing through the filters 321 to 32n and 331 to 33n (see FIG. 1).

  The arithmetic circuit 35b subtracts the audio signal Y141 of the microphone M1 from the audio signal Y221 of the microphone M2, thereby generating an audio signal with reduced audio components from the speaker SP over the frequency bands G1 to Gn. The audio signals Y141 and Y221 for the sound emitted by the SP have the same phase and amplitude in the frequency band G1, and in particular, an audio signal in which the audio component from the speaker SP in the frequency band G1 is canceled is generated. Similarly, by subtracting the sound signal Y142 of the microphone M1 from the sound signal Y222 of the microphone M2, a sound signal in which the sound component from the speaker SP is canceled in the frequency band G2 is generated. By subtracting the audio signal Y14n of the microphone M1 from the audio signal Y22n, an audio signal in which the audio component from the speaker SP is canceled in the frequency band Gn is generated, and all the subtraction results in each frequency band G1 to Gn are generated. By adding and dividing the addition result by the frequency band division number n, an audio signal Ya in which the audio component from the speaker SP is canceled for each frequency band G1 to Gn is generated.

  According to the averaging process, it is possible to obtain substantially the same effect as the frequency division of the first embodiment without providing the bandpass filters 321-32n and 331-33n filter means. Detailed description.

  In the averaging process, the difference between the sound signals of the microphones M1 and M2 whose phases and amplitudes are matched for the frequency bands G1 to Gn with respect to the sound emitted from the speaker SP is separated into the frequency bands G1 to Gn. This is a process of simply adding without passing through the filter means, and dividing and averaging by dividing the frequency band by the division number n. The filter means is not required, and the circuit scale, cost and size can be reduced. . The following [Equation 10] is a mathematical expression for deriving the speaker sound cancellation amount Zmave by the averaging process.

Here, Zmk is the amount of cancellation of speaker sound by the above subtraction processing of the sound signals of the microphones M1 and M2 whose phases and amplitudes are matched in the frequency band Gk. Specifically, Zm1 is the phase in the frequency band G1. , The amount of speaker sound cancellation by the subtraction processing of the audio signals of the microphones M1 and M2 having the same amplitude, and Zm2 is the subtraction of the audio signals of the microphones M1 and M2 having the same phase and amplitude in the frequency band G2. Zmn is the amount of speaker sound canceled by the above subtraction processing of the sound signals of the microphones M1 and M2 whose phases and amplitudes are matched in the frequency band Gn.

  For example, as shown in FIG. 18, the speaker sound cancellation amounts Ys1 and Ys2 when the speaker sound cancellation amounts are adjusted to be optimal at frequencies of 1 KHz and 3 KHz, respectively, are added, and the result of the addition is added to the frequency band. By dividing by the division number n = 2, the theoretical value Yt21 of the cancellation amount by the averaging process of the present embodiment is obtained.

  Further, FIG. 19 is divided into a frequency band G1 of 2 KHz or less optimized for a speaker sound cancellation amount at a frequency of 1 KHz, and a frequency band G2 of 2 KHz or more optimized for a speaker sound cancellation amount of a frequency of 3 KHz. The measured value Ys21 of the speaker sound cancellation amount when the averaging process is performed, and the approximate curve Ys22 of the actual measurement value Ys21 are shown, and there is no frequency band with an extremely high cancellation amount, but the cancellation is performed in a wide frequency band. It can be seen that the amount is averaged and there is a sufficient speaker sound canceling effect over a wide frequency band.

  In the present embodiment, one A / D conversion circuit is provided on the microphone M1 side, a plurality of A / D conversion circuits are provided on the microphone M2, and the A / D conversion circuit on the microphone M1 side performs A / D conversion. The A / D conversion circuit on the microphone M2 side performs A / D conversion at the timing after the delay times Td1 to Tdn for the frequency bands G1 to Gn from the timing of performing the audio signal, so that the audio signal of the microphone M1 and the microphone M2 Each phase of the audio signal is matched for each of the frequency bands G1 to Gn. However, contrary to this configuration, a plurality of A / D conversion circuits are provided on the microphone M1 side, one A / D conversion circuit is provided on the microphone M2, and the A / D conversion circuit on the microphone M2 side is an A / D conversion circuit. The A / D conversion circuit on the microphone M1 side performs A / D conversion at the timing before the delay time Td1 to Tdn for each frequency band G1 to Gn from the timing of performing D conversion, and the above arithmetic processing is performed on these digital signals. The same effect can be obtained by applying.

2 is a circuit configuration diagram of a signal processing unit of the communication device according to Embodiment 1. FIG. It is side surface sectional drawing which shows a structure same as the above. It is a perspective view which shows a structure same as the above. It is a circuit diagram which shows the structure of an audio | voice processing part same as the above. It is side surface sectional drawing which shows the structure of a microphone substrate same as the above. It is side surface sectional drawing which shows the structure of a bare chip same as the above. It is the (a) simplified top view and (b) simplified circuit diagram which show the structure of a microphone substrate same as the above. It is a circuit diagram of an impedance conversion circuit same as the above. (A)-(h) It is a signal waveform diagram of the signal processing part same as the above. (A) (b) It is a signal waveform diagram of a signal processing part same as the above. (A) (b) It is a signal waveform diagram of a signal processing part same as the above. (A) (b) It is a signal waveform diagram of a signal processing part same as the above. It is a phasor vector figure of the residual signal from a speaker same as the above. (A) Amplitude ratio of speaker sound actually measured, and (b) Frequency characteristics of phase difference of speaker sound are shown. (A) The theoretical value of the amount of cancellation of speaker sound when frequency division is not performed, and (b) Each frequency characteristic of the measured value is shown. (A) The theoretical value of the amount of cancellation of the speaker sound when frequency division is performed, and (b) the frequency characteristics of the actually measured value. It is a circuit block diagram of the signal processing part of the telephone apparatus of Embodiment 2. The frequency characteristic of the theoretical value of the cancellation amount by the averaging process is shown. The frequency characteristic of the actual measurement value of the cancellation amount by the averaging process is shown.

Explanation of symbols

A Caller SP Speaker M1, M2 Microphone 10e Signal processing unit 301 A / D conversion circuit 311 to 31n A / D conversion circuit 321 to 32n Band pass filter 331 to 33n Band pass filter 341 to 34n Attenuation circuit 35a Arithmetic circuit 36 Timing control Part

Claims (4)

A speaker that outputs the transmitted sound information, a first microphone that collects sound and outputs an analog sound signal, and a distance from the speaker that is farther than the first microphone, collects sound. A second microphone for outputting an analog audio signal, and a signal processing unit for processing and transmitting each of the audio signals output from the first and second microphones. The transmission time of the sound wave corresponding to the difference in distance between the microphone and the speaker is the delay time,
The signal processing unit includes a plurality of any one of first and second A / D conversion means for converting analog audio signals output from the first and second microphones into digital signals, respectively, The / D conversion means respectively correspond to mutually different predetermined frequency bands, and each A / D conversion is performed at a timing shifted by a delay time for each predetermined frequency band from the timing at which the other A / D conversion means performs A / D conversion. Level adjusting means for performing processing for matching the output levels of the first and second microphones for the sound from the speaker for each predetermined frequency band on the digital signal output by the A / D conversion means, and level adjustment And a calculating means for outputting a difference between the audio signals of the first and second microphones that have passed through the means.   The signal processing unit separates the output of the first filter unit that passes a predetermined frequency band for each output of the one A / D conversion unit and the output of the other A / D conversion unit for each predetermined frequency band. And a second filter means that passes the first and second filter means and the level adjusting means for each predetermined frequency band of the audio signals of the first and second microphones that have passed through the level adjusting means. The communication device according to claim 1, wherein the difference is added.   The signal processing unit includes one first A / D conversion unit, a plurality of second A / D conversion units, and a plurality of second A / D conversion units configured to have a predetermined frequency band. Corresponding to the first A / D conversion, the A / D conversion is performed at the timing after the delay time for each predetermined frequency band from the timing at which the first A / D conversion means performs the A / D conversion, and the first A / D conversion is further performed. First filter means for allowing the output of the conversion means to pass separately for each predetermined frequency band, and second filter means for passing the predetermined frequency band for each output of the second A / D conversion means. The calculating means adds a difference for each predetermined frequency band of the audio signals of the first and second microphones that have passed through the first and second filter means and the level adjusting means. 3. The communication device according to 1 or 2.   The signal processing unit includes a plurality of the first A / D conversion units, includes one second A / D conversion unit, and the plurality of first A / D conversion units include a predetermined frequency band. Corresponding to each of the above, the A / D conversion is performed at a timing before the delay time for each predetermined frequency band from the timing at which the second A / D conversion means performs the A / D conversion, and the first A / D conversion is further performed. First filter means for allowing a predetermined frequency band to pass for each output of the conversion means, and second filter means for allowing the output of the second A / D conversion means to be separated and passed for each predetermined frequency band. The calculating means adds a difference for each predetermined frequency band of the audio signals of the first and second microphones that have passed through the first and second filter means and the level adjusting means. 3. The communication device according to 1 or 2.

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